227 lines
5.6 KiB
Go
227 lines
5.6 KiB
Go
package webrtc
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import (
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"fmt"
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"time"
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"github.com/pion/webrtc/v2"
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"github.com/pkg/errors"
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"github.com/rs/zerolog"
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"github.com/rs/zerolog/log"
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"n.eko.moe/neko/internal/config"
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"n.eko.moe/neko/internal/event"
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"n.eko.moe/neko/internal/gst"
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"n.eko.moe/neko/internal/hid"
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"n.eko.moe/neko/internal/message"
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"n.eko.moe/neko/internal/session"
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)
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func New(sessions *session.SessionManager, conf *config.WebRTC) *WebRTCManager {
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logger := log.With().Str("module", "webrtc").Logger()
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engine := webrtc.MediaEngine{}
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engine.RegisterDefaultCodecs()
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setings := webrtc.SettingEngine{
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LoggerFactory: loggerFactory{
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logger: logger,
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},
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}
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return &WebRTCManager{
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logger: logger,
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engine: engine,
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setings: setings,
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api: webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(setings)),
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cleanup: time.NewTicker(1 * time.Second),
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shutdown: make(chan bool),
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sessions: sessions,
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conf: conf,
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config: webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{
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{
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URLs: []string{"stun:stun.l.google.com:19302"},
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},
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},
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SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
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},
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}
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}
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type WebRTCManager struct {
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logger zerolog.Logger
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engine webrtc.MediaEngine
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setings webrtc.SettingEngine
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config webrtc.Configuration
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sessions *session.SessionManager
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api *webrtc.API
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video *webrtc.Track
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audio *webrtc.Track
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videoPipeline *gst.Pipeline
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audioPipeline *gst.Pipeline
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cleanup *time.Ticker
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conf *config.WebRTC
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shutdown chan bool
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}
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func (m *WebRTCManager) Start() {
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hid.Display(m.conf.Display)
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switch m.conf.VideoCodec {
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case "vp8":
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if err := m.createVideoTrack(webrtc.DefaultPayloadTypeVP8); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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case "vp9":
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if err := m.createVideoTrack(webrtc.DefaultPayloadTypeVP9); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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case "h264":
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if err := m.createVideoTrack(webrtc.DefaultPayloadTypeH264); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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default:
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m.logger.Panic().Err(errors.Errorf("unknown video codec %s", m.conf.AudioCodec)).Msg("unable to start webrtc manager")
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}
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switch m.conf.AudioCodec {
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case "opus":
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if err := m.createAudioTrack(webrtc.DefaultPayloadTypeOpus); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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case "g722":
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if err := m.createAudioTrack(webrtc.DefaultPayloadTypeG722); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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case "pcmu":
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if err := m.createAudioTrack(webrtc.DefaultPayloadTypePCMU); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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case "pcma":
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if err := m.createAudioTrack(webrtc.DefaultPayloadTypePCMA); err != nil {
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m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
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}
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default:
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m.logger.Panic().Err(errors.Errorf("unknown audio codec %s", m.conf.AudioCodec)).Msg("unable to start webrtc manager")
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}
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m.videoPipeline.Start()
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m.audioPipeline.Start()
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go func() {
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defer func() {
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m.logger.Info().Msg("shutdown")
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}()
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for {
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select {
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case <-m.shutdown:
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return
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case <-m.cleanup.C:
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hid.Check(time.Second * 10)
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}
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}
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}()
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m.sessions.OnHostCleared(func(id string) {
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hid.Reset()
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})
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m.sessions.OnCreated(func(id string, session *session.Session) {
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m.logger.Debug().Str("id", id).Msg("session created")
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})
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m.sessions.OnDestroy(func(id string) {
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m.logger.Debug().Str("id", id).Msg("session destroyed")
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})
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// TODO: log resolution, bit rate and codec parameters
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m.logger.Info().
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Str("video_display", m.conf.Display).
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Str("video_codec", m.conf.VideoCodec).
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Str("audio_device", m.conf.Device).
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Str("audio_codec", m.conf.AudioCodec).
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Msgf("webrtc streaming")
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}
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func (m *WebRTCManager) Shutdown() error {
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m.logger.Info().Msgf("webrtc shutting down")
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m.cleanup.Stop()
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m.shutdown <- true
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m.videoPipeline.Stop()
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m.audioPipeline.Stop()
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return nil
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}
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func (m *WebRTCManager) CreatePeer(id string, sdp string) error {
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session, ok := m.sessions.Get(id)
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if !ok {
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return fmt.Errorf("invalid session id %s", id)
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}
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peer, err := m.api.NewPeerConnection(m.config)
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if err != nil {
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return err
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}
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if _, err := peer.AddTransceiverFromTrack(m.video, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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}); err != nil {
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return err
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}
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if _, err := peer.AddTransceiverFromTrack(m.audio, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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}); err != nil {
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return err
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}
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peer.SetRemoteDescription(webrtc.SessionDescription{
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SDP: sdp,
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Type: webrtc.SDPTypeOffer,
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})
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answer, err := peer.CreateAnswer(nil)
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if err != nil {
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return err
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}
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if err = peer.SetLocalDescription(answer); err != nil {
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return err
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}
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if err := session.Send(message.Signal{
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Event: event.SIGNAL_ANSWER,
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SDP: answer.SDP,
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}); err != nil {
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return err
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}
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peer.OnDataChannel(func(d *webrtc.DataChannel) {
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d.OnMessage(func(msg webrtc.DataChannelMessage) {
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if err = m.handle(id, msg); err != nil {
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m.logger.Warn().Err(err).Msg("data handle failed")
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}
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})
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})
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peer.OnConnectionStateChange(func(connectionState webrtc.PeerConnectionState) {
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switch connectionState {
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case webrtc.PeerConnectionStateDisconnected:
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case webrtc.PeerConnectionStateFailed:
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m.logger.Info().Str("id", id).Msg("peer disconnected")
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m.sessions.Destroy(id)
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break
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case webrtc.PeerConnectionStateConnected:
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m.logger.Info().Str("id", id).Msg("peer connected")
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break
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}
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})
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m.sessions.SetPeer(id, peer)
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return nil
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}
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